2 lines With 1 Phone Set

[kraza01]

Hello there,
is there a way where I may be able to use my two VOIP lines, Line 1 (with FPL) and Line 2 (with a different provider) using only 1 phone set and one ATA (I have SPA 2102 and HT 502).

[FONGO_steve]

www.sipsorcery.com

I do the same thing. I have DIDs in two different cities through freephoneline. Sip Sorcery handles all the routing and they both ring one handset. I can even set which number to call out from based on different numbers called. There's some info in our device configuration section here about it.

[kraza01]

Can we trust www.sipsorcery.com, as we are providing our Sip settings there.

[FONGO_kris]

Yes you can... I'm using it, I have no problem.

[FONGO_steve]

I looked into it quite a bit and they seem pretty reliable and trustworthy. It's basically the old mysipswitch.com service that a lot of people used to use.
Of course it's entirely your call whether or not you trust them or not - there is also the element of risk when providing your account credentials to a third-party service. When I have the time, I plan on building an Asterisk box to basically do what SIP Sorcery does.

[Mysterious]

i did try yesterday , created acount with sip sourcery and put all my FPl and one other provider , but for some reason both FPL and Inphonex sip account did not went through means its not saying registetred, then i tried to put the sip sourcery account on Softphone to see it works , yes it was logging but again nothing else happened , any help will be appreciated, further i put the settings as its mentioned in device configuration thread by Steve.

[kraza01]

Thnaks for the initial reply, Now i am able to dial out successfully usuing my FPL as well as V buzz account via sipsourcery but i need help to get incoming calls as right now all my calls are going to straight Voice mail for both account.
Any help will be appreciated.
Again I am not able to get incoming calls via sipsourcery,

[FONGO_kris]

Have you looked at Steves SipSorcery guide in Device Config..? --> http://forum.freephoneline.ca/viewtopic.php?f=15&t=429

[kraza01]

yes that was really help full and i used all info according to that , but for some reason (it may be the dial plan where i am confuse, all i want all calls to the one phone line ) so i wonder if Steve can write that one dial plan where i can see what i am missing in mine , as i am able to dial out perfectly but all call to me are going to vmail. diai plan only for Incoming

[FONGO_kris]

Steve is checking his account right now and will probably respond shortly

[FONGO_steve]

Hey Kraza,
Have you tried the Ruby Wizard for creating your dial-plan?
http://www.mysipswitch.com/rubyzard.aspx

Make sure you use the same naming convention in sipsorcery as you do in the ruby wizard (as they state in the instructions). This should allow you to assign how you want the numbers to be used for incoming/outgoing calls.

If that doesn't work, I can take my dial-plan that I made in ruby and attempt to break it down a little further for you to edit.

[kraza01]

Thanks Again for reply
here my scenario is 2 lines FPL and Vbuzz and i get this plan usuing that wizard, so where u think i need to change manually as after using this my outgoing is fine but may be some thing is missing in incoming portion. rest i sued the naming exactly as they are in sip sourcery.


#Ruby
# Dial Plan Generated by Rubyzard v0.1
# If you need help, please post in our forum
# http://www.mysipswitch.com

# SIP tracing : true or false
sys.Trace = false

sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")

if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
sys.Respond(480, "#{sys.Username} Not available")
else
sys.Dial("Enter Number@FPL",30)
sys.Respond(480, "#{sys.Username} Not available")
end

else
# Do your OUTGOING call processing customisations here.
sys.Dial("FPL")

end

[kraza01]

Hi Steve i will look forward for your demo dial plan for incoming calls.
Thanks

[FONGO_steve]

I'll be honest - your dial string is exactly the same as what I am using. The only difference was what you named the services - but this won't affect a thing.

What sort of error occurs when you call into your number? Do you get any sort of verbal dialogue, or just dead air?

[kraza01]

My both providers are only register at sip sourcery and its dialing out fine, only problem is all incoming calls going straight to v mail , voice mail is not activated either of the line from their respective web control panels . and so far i can only suspect my incoming dial plan portion.

[kraza01]

Hello there,
Do i need to replace any info from the following dial plan (like replace any of the portion with my sip sourcery sip log in name etc) as so far i have not done any changes in it and that might be the reason i am not getting incoming calls. will look forward for any response. Thanks.
Dail plan for Incming only (as my out going portion is 100% ok)

# SIP tracing : true or false
sys.Trace = false

sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")

if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
sys.Respond(480, "#{sys.Username} Not available")
else

[FONGO_steve]

Hello there,
Do i need to replace any info from the following dial plan (like replace any of the portion with my sip sourcery sip log in name etc) as so far i have not done any changes in it and that might be the reason i am not getting incoming calls. will look forward for any response. Thanks.
Dail plan for Incming only (as my out going portion is 100% ok)

# SIP tracing : true or false
sys.Trace = false

sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")

if sys.In then
# Do your INCOMING call processing customisations here.
if sys.IsAvailable() then
sys.Dial("#{sys.Username}@local",30)
sys.Respond(480, "#{sys.Username} Not available")
else

If outgoing calls work then we know that registration is good. I'm not a pro at understanding the ruby coding though - your best bet would probably be asking in the sipsorcery forums. I don't want you to wait around forever for me to try and figure this out as I'm a pretty busy guy during the day here

[kraza01]

Finally my incoming calls are working fine if i run sipsorcery on HT502, but this is not The End of mystery , at HT502 my dial out did not worked, so if i want dial out its working on SPA2102 with no problem (same set up and same dial plan etc on both devices adn at sipsorcery) and my dial in are working perfect with HT502, no clue why its happening so ,
But I am happy at least i am able to get the incoming calls, dose no matter i have to use my back up ATA HT502.